| /* | |
| * Copyright 2019 The WebRTC Project Authors. All rights reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef API_PACKET_SOCKET_FACTORY_H_ | |
| #define API_PACKET_SOCKET_FACTORY_H_ | |
| #include <memory> | |
| #include <string> | |
| #include <vector> | |
| #include "api/async_dns_resolver.h" | |
| #include "api/wrapping_async_dns_resolver.h" | |
| #include "rtc_base/async_packet_socket.h" | |
| #include "rtc_base/proxy_info.h" | |
| #include "rtc_base/system/rtc_export.h" | |
| namespace rtc { | |
| class SSLCertificateVerifier; | |
| class AsyncResolverInterface; | |
| struct PacketSocketTcpOptions { | |
| PacketSocketTcpOptions() = default; | |
| ~PacketSocketTcpOptions() = default; | |
| int opts = 0; | |
| std::vector<std::string> tls_alpn_protocols; | |
| std::vector<std::string> tls_elliptic_curves; | |
| // An optional custom SSL certificate verifier that an API user can provide to | |
| // inject their own certificate verification logic (not available to users | |
| // outside of the WebRTC repo). | |
| SSLCertificateVerifier* tls_cert_verifier = nullptr; | |
| }; | |
| class RTC_EXPORT PacketSocketFactory { | |
| public: | |
| enum Options { | |
| OPT_STUN = 0x04, | |
| // The TLS options below are mutually exclusive. | |
| OPT_TLS = 0x02, // Real and secure TLS. | |
| OPT_TLS_FAKE = 0x01, // Fake TLS with a dummy SSL handshake. | |
| OPT_TLS_INSECURE = 0x08, // Insecure TLS without certificate validation. | |
| // Deprecated, use OPT_TLS_FAKE. | |
| OPT_SSLTCP = OPT_TLS_FAKE, | |
| }; | |
| PacketSocketFactory() = default; | |
| virtual ~PacketSocketFactory() = default; | |
| virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address, | |
| uint16_t min_port, | |
| uint16_t max_port) = 0; | |
| virtual AsyncListenSocket* CreateServerTcpSocket( | |
| const SocketAddress& local_address, | |
| uint16_t min_port, | |
| uint16_t max_port, | |
| int opts) = 0; | |
| virtual AsyncPacketSocket* CreateClientTcpSocket( | |
| const SocketAddress& local_address, | |
| const SocketAddress& remote_address, | |
| const ProxyInfo& proxy_info, | |
| const std::string& user_agent, | |
| const PacketSocketTcpOptions& tcp_options) = 0; | |
| // The AsyncResolverInterface is deprecated; users are encouraged | |
| // to switch to the AsyncDnsResolverInterface. | |
| // TODO(bugs.webrtc.org/12598): Remove once all downstream users | |
| // are converted. | |
| virtual AsyncResolverInterface* CreateAsyncResolver() { | |
| // Default implementation, so that downstream users can remove this | |
| // immediately after changing to CreateAsyncDnsResolver | |
| RTC_DCHECK_NOTREACHED(); | |
| return nullptr; | |
| } | |
| virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> | |
| CreateAsyncDnsResolver() { | |
| // Default implementation, to aid in transition to AsyncDnsResolverInterface | |
| return std::make_unique<webrtc::WrappingAsyncDnsResolver>( | |
| CreateAsyncResolver()); | |
| } | |
| private: | |
| PacketSocketFactory(const PacketSocketFactory&) = delete; | |
| PacketSocketFactory& operator=(const PacketSocketFactory&) = delete; | |
| }; | |
| } // namespace rtc | |
| #endif // API_PACKET_SOCKET_FACTORY_H_ |